- 13 4月, 2012 3 次提交
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由 Takashi Iwai 提交于
Many fields have been moved to struct snd_usb_endpoint. Also fix the proc output to correspond to the new structure. Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Daniel Mack 提交于
With the previous commit that added the new streaming model, all endpoint and streaming related code is now in endpoint.c, and pcm.c only acts as a wrapper for handling the packet's payload. Signed-off-by: NDaniel Mack <zonque@gmail.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Daniel Mack 提交于
This patch adds a new generic streaming logic for audio over USB. It defines a model (snd_usb_endpoint) that handles everything that is related to an USB endpoint and its streaming. There are functions to activate and deactivate an endpoint (which call usb_set_interface()), and to start and stop its URBs. It also has function pointers to be called when data was received or is about to be sent, and pointer to a sync slave (another snd_usb_endpoint) that is informed when data has been received. A snd_usb_endpoint knows about its state and implements a refcounting, so only the first user will actually start the URBs and only the last one to stop it will tear them down again. With this sort of abstraction, the actual streaming is decoupled from the pcm handling, which makes the "implicit feedback" mechanisms easy to implement. In order to split changes properly, this patch only adds the new implementation but leaves the old one around, so the the driver doesn't change its behaviour. The switch to actually use the new code is submitted separately. Signed-off-by: NDaniel Mack <zonque@gmail.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 15 2月, 2012 1 次提交
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由 Xi Wang 提交于
A malicious USB device could feed in a large nr_rates value. This would cause the subsequent call to kmemdup() to allocate a smaller buffer than expected, leading to out-of-bounds access. This patch validates the nr_rates value and reuses the limit introduced in commit 4fa0e81b ("ALSA: usb-audio: fix possible hang and overflow in parse_uac2_sample_rate_range()"). Signed-off-by: NXi Wang <xi.wang@gmail.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 12 9月, 2011 1 次提交
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由 Pierre-Louis Bossart 提交于
Existing code only updates the audio delay when URBs were submitted/retired. This can introduce an uncertainty of 8ms on the number of samples played out with the default settings, and a lot more when URBs convey more packets to reduce the interrupt rate and power consumption. This patch relies on the USB frame counter to reduce the uncertainty to less than 2ms worst-case. The delay information essentially becomes independent of the URB size and number of packets. This should help applications like PulseAudio which require accurate audio timing. Clemens Ladisch reported a decrease of mplayer's A-V difference from nrpacks down to at most 1ms. Thanks to Clemens for also pointing out that the implementation of frame counters varies between different HCDs. Only the 8 lowest-bits are used to estimate the delay. Signed-off-by: NPierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> [clemens: changed debug code] Signed-off-by: NClemens Ladisch <clemens@ladisch.de> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 27 10月, 2010 1 次提交
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由 Clemens Ladisch 提交于
There are two USB Audio Class specifications (v1 and v2), but neither of them clearly defines the feedback format for high-speed UAC v1 devices. Add to this whatever the Creative and M-Audio firmware writers have been smoking, and it becomes impossible to predict the exact feedback format used by a particular device. Therefore, automatically detect the feedback format by looking at the magnitude of the first received feedback value. Also, this allows us to get rid of some special cases for E-Mu devices. Signed-off-by: NClemens Ladisch <clemens@ladisch.de> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 01 6月, 2010 1 次提交
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由 Daniel Mack 提交于
Audio devices which comply to the UAC2 standard can export complex clock topologies in its descriptors and set up links between them. The entities that are defined are - clock sources, which define the end-leafs. - clock selectors, which act as switch to select one out of many possible clocks sources. - clock multipliers, which have an input clock source, and act as clock source again. They can be used to derive one clock from another. All sample rate changes, clock validity queries and the like must go to clock source elements, while clock selectors and multipliers can be used as terminal clock source. The following patch adds a parser for these elements and functions to iterate over the tree and find the leaf nodes (clock sources). The samplerate set functions were moved to the new clock.c file. Signed-off-by: NDaniel Mack <daniel@caiaq.de> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 05 3月, 2010 3 次提交
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由 Clemens Ladisch 提交于
In preparation for USB audio 2.0 support, change the audioformat structure so that it uses a bitmask to specify possible formats. Signed-off-by: NClemens Ladisch <clemens@ladisch.de> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Clemens Ladisch 提交于
The snd_usb_substream::format field actually contains the index of the current alternate setting, so rename it to altset_idx to avoid confusion. Signed-off-by: NClemens Ladisch <clemens@ladisch.de> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Daniel Mack 提交于
Clean up the usb audio driver by factoring out a lot of functions to separate files. Code for procfs, quirks, urbs, format parsers etc all got a new home now. Moved almost all special quirk handling to quirks.c and introduced new generic functions to handle them, so the exceptions do not pollute the whole driver. Renamed usbaudio.c to card.c because this is what it actually does now. Renamed usbmidi.c to midi.c for namespace clarity. Removed more things from usbaudio.h. The non-standard drivers were adopted accordingly. Signed-off-by: NDaniel Mack <daniel@caiaq.de> Cc: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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